Notes:
 This parameter can only be configured for an IP Profile using the
IPProfile parameter (see 'Configuring IP Profiles' on page 196).
 For IP-to-IP calls, you can configure the device to route calls to an
alternative IP Group when the maximum number of concurrent calls
is reached. To do so, you need to add an alternative routing rule in
the Outbound IP Routing table that reroutes the call to an alternative
IP Group. You also need to add a rule to the Reason for Alternative
Routing table to initiate an alternative rule for Tel-to-IP calls using
cause 805.
Web: QoS statistics in SIP
Release Call
[QoSStatistics]
Enables the device to include call quality of service (QoS) statistics in
SIP BYE and SIP 200 OK response to BYE, using the proprietary SIP
header X-RTP-Stat.
 [0] = Disable (default)
 [1] = Enable
The X-RTP-Stat header provides the following statistics:
 Number of received and sent voice packets
 Number of received and sent voice octets
 Received packet loss, jitter (in ms), and latency (in ms)
The X-RTP-Stat header contains the following fields:
 PS=<voice packets sent>
 OS=<voice octets sent>
 PR=<voice packets received>
 OR=<voice octets received>
 PL=<receive packet loss>
 JI=<jitter in ms>
 LA=<latency in ms>
Below is an example of the X-RTP-Stat header in a SIP BYE message:
BYE sip:302@10.33.4.125 SIP/2.0
Via: SIP/2.0/UDP
10.33.4.126;branch=z9hG4bKac2127550866
Max-Forwards: 70
From:
<sip:401@10.33.4.126;user=phone>;tag=1c2113553324
To: <sip:302@company.com>;tag=1c991751121
Call-ID: 991750671245200001912@10.33.4.125
CSeq: 1 BYE
X-RTP-Stat:
PS=207;OS=49680;;PR=314;OR=50240;PL=0;JI=600;LA=40;
Supported: em,timer,replaces,path,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK
,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Sip-Gateway-/v.6.2A.008.006
Reason: Q.850 ;cause=16 ;text="local"
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